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<tt>Hi Nick,<br>
<br>
The BYE is not properly formed and rejected by script - in the 200
OK of the INVITE, you can see that your opensips is doing
Record-Routing, but the BYE does not contain the corresponding
Route hdr, so SIP routing is impossible.<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<br>
On 04/09/2013 08:05 PM, Nick Khamis wrote:
<blockquote
cite="mid:CAGWRaZb-rNa=sGYTfOY48YTi4yEzeRwO59GzWRHpr3V0ZUyRpw@mail.gmail.com"
type="cite">
<div dir="ltr">Hello Everyone,
<div><br>
</div>
<div style="">I saw an earlier post about this issue: <a
moz-do-not-send="true"
href="http://www.mail-archive.com/users@lists.opensips.org/msg23052.html">http://www.mail-archive.com/users@lists.opensips.org/msg23052.html</a></div>
<div style=""><br>
</div>
<div style="">And was wondering if there was anything we can do
on our end to fix this problem? It seems that providers are
not obligated to maintain RR? When the caller (internal)
initiates the BYE everything is ok, but not the case when the
callee (external) initiates the BYE.</div>
<div style=""><br>
</div>
<div style=""><a moz-do-not-send="true"
href="http://192.168.2.5">192.168.2.5</a>: OpenSIPS</div>
<div style=""><a moz-do-not-send="true"
href="http://192.168.2.10">192.168.2.10</a>: Asterisk</div>
<div style=""><a moz-do-not-send="true"
href="http://70.10.163.44">70.10.163.44</a>: Public IP<br>
</div>
<div style=""><a moz-do-not-send="true"
href="http://108.59.2.133">108.59.2.133</a>: Service
Provider<br>
</div>
<div style=""><br>
</div>
<div style=""><br>
</div>
<div style="">
<div>U 2013/04/09 12:17:02.920454 <a moz-do-not-send="true"
href="http://192.168.2.10:5060">192.168.2.10:5060</a>
-> <a moz-do-not-send="true"
href="http://192.168.2.5:5060">192.168.2.5:5060</a></div>
<div>SIP/2.0 200 OK.</div>
<div>Via: SIP/2.0/UDP
192.168.2.5;branch=z9hG4bKac2e.554c6e93.0;received=192.168.2.5;rport=5060.</div>
<div>Via: SIP/2.0/UDP
192.168.2.11:5060;rport=5060;received=192.168.2.11;branch=z9hG4bK42f3f16e7BC15DF1.</div>
<div>Record-Route:
<a class="moz-txt-link-rfc2396E" href="sip:192.168.2.5;lr;did=392.62562fb2"><sip:192.168.2.5;lr;did=392.62562fb2></a>.</div>
<div>From: "1001" <<a moz-do-not-send="true"
href="mailto:sip%3A1001@server.example.com">sip:1001@server.example.com</a>>;tag=FCA0BFC0-B585477D.</div>
<div>To: <<a moz-do-not-send="true"
href="mailto:sip%3A15178342008@server.example.com">sip:15178342008@server.example.com</a>;user=phone>;tag=as0a76fcde.</div>
<div>Call-ID: <a moz-do-not-send="true"
href="mailto:595ad334-f06e97fa-3bbc8137@192.168.2.11">595ad334-f06e97fa-3bbc8137@192.168.2.11</a>.</div>
<div>CSeq: 1 INVITE.</div>
<div>Server: Asterisk PBX UNKNOWN__and_probably_unsupported.</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH.</div>
<div>Supported: replaces, timer.</div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:15178342008@192.168.2.10:5060">sip:15178342008@192.168.2.10:5060</a>>.</div>
<div>Content-Type: application/sdp.</div>
<div>Content-Length: 312.</div>
<div>.</div>
<div>v=0.</div>
<div>o=root 1860889533 1860889534 IN IP4 192.168.2.10.</div>
<div>s=Asterisk PBX UNKNOWN__and_probably_unsupported.</div>
<div>c=IN IP4 192.168.2.10.</div>
<div>t=0 0.</div>
<div>m=audio 60646 RTP/AVP 18 101.</div>
<div>a=rtpmap:18 G729/8000.</div>
<div>a=fmtp:18 annexb=no.</div>
<div>a=rtpmap:101 telephone-event/8000.</div>
<div>a=fmtp:101 0-16.</div>
<div>a=silenceSupp:off - - - -.</div>
<div>a=ptime:20.</div>
<div>a=sendrecv.</div>
<div>
<br>
</div>
<div>ACC: transaction answered:
timestamp=1365524222;method=INVITE;from_tag=FCA0BFC0-B585477D;to_tag=as0a76fcde;call_id=<a
moz-do-not-send="true"
href="mailto:595ad334-f06e97fa-3bbc8137@192.168.2.11">595ad334-f06e97fa-3bbc8137@192.168.2.11</a>;code=200;reason=OK</div>
<div><br>
</div>
<div>U 2013/04/09 12:17:02.939608 <a moz-do-not-send="true"
href="http://192.168.2.5:5060">192.168.2.5:5060</a> ->
<a moz-do-not-send="true" href="http://192.168.2.11:5060">192.168.2.11:5060</a></div>
<div>SIP/2.0 200 OK.</div>
<div>Via: SIP/2.0/UDP
192.168.2.11:5060;rport=5060;received=192.168.2.11;branch=z9hG4bK42f3f16e7BC15DF1.</div>
<div>Record-Route:
<a class="moz-txt-link-rfc2396E" href="sip:192.168.2.5;lr;did=392.62562fb2"><sip:192.168.2.5;lr;did=392.62562fb2></a>.</div>
<div>From: "1001" <<a moz-do-not-send="true"
href="mailto:sip%3A1001@server.example.com">sip:1001@server.example.com</a>>;tag=FCA0BFC0-B585477D.</div>
<div>To: <<a moz-do-not-send="true"
href="mailto:sip%3A15178342008@server.example.com">sip:15178342008@server.example.com</a>;user=phone>;tag=as0a76fcde.</div>
<div>Call-ID: <a moz-do-not-send="true"
href="mailto:595ad334-f06e97fa-3bbc8137@192.168.2.11">595ad334-f06e97fa-3bbc8137@192.168.2.11</a>.</div>
<div>CSeq: 1 INVITE.</div>
<div>Server: Asterisk PBX UNKNOWN__and_probably_unsupported.</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH.</div>
<div>Supported: replaces, timer.</div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:15178342008@192.168.2.10:5060">sip:15178342008@192.168.2.10:5060</a>>.</div>
<div>Content-Type: application/sdp.</div>
<div>Content-Length: 329.</div>
<div>.</div>
<div>v=0.</div>
<div>o=root 1860889533 1860889534 IN IP4 192.168.2.10.</div>
<div>s=Asterisk PBX UNKNOWN__and_probably_unsupported.</div>
<div>c=IN IP4 192.168.2.5.</div>
<div>t=0 0.</div>
<div>m=audio 31148 RTP/AVP 18 101.</div>
<div>a=rtpmap:18 G729/8000.</div>
<div>a=fmtp:18 annexb=no.</div>
<div>a=rtpmap:101 telephone-event/8000.</div>
<div>a=fmtp:101 0-16.</div>
<div>a=silenceSupp:off - - - -.</div>
<div>a=ptime:20.</div>
<div>a=sendrecv.</div>
<div>
a=nortpproxy:yes.</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>U 2013/04/09 12:17:06.988918 <a moz-do-not-send="true"
href="http://108.59.2.133:5060">108.59.2.133:5060</a>
-> <a moz-do-not-send="true"
href="http://192.168.2.5:5060">192.168.2.5:5060</a></div>
<div>BYE <a moz-do-not-send="true"
href="http://sip:1001@70.10.163.44:5060">sip:1001@70.10.163.44:5060</a>
SIP/2.0.</div>
<div>Max-Forwards: 64.</div>
<div>To: "1001" <<a moz-do-not-send="true"
href="mailto:sip%3A1001@70.10.163.44">sip:1001@70.10.163.44</a>>;tag=as4b40d9b4.</div>
<div>From: <<a moz-do-not-send="true"
href="mailto:sip%3A001110215178342008@sbc.voxbeam.com">sip:001110215178342008@sbc.voxbeam.com</a>>;tag=3574513019-870807.</div>
<div>Reason: Q.850;cause=16;text="".</div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060">705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060</a>.</div>
<div>CSeq: 2 BYE.</div>
<div>Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER,
NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE.</div>
<div>Via: SIP/2.0/UDP
108.59.2.133;branch=z9hG4bK2deb.8bfd0b06.0.</div>
<div>Contact: <<a moz-do-not-send="true"
href="mailto:sip%3Acallee@108.59.2.133">sip:callee@108.59.2.133</a>;did=e9e.a6618961>.</div>
<div>Allow-Events: as-feature-event.</div>
<div>Allow-Events: call-info.</div>
<div>Allow-Events: presence.</div>
<div>Allow-Events: line-seize.</div>
<div>Allow-Events: dialog.</div>
<div>Allow-Events: refer.</div>
<div>Allow-Events: message-summary.</div>
<div>Content-Length: 0.</div>
<div>.</div>
<div><br>
</div>
<div>Forcing RPORT: <a moz-do-not-send="true"
href="mailto:sip%3A001110215178342008@sbc.voxbeam.com">sip:001110215178342008@sbc.voxbeam.com</a></div>
<div><br>
</div>
<div>U 2013/04/09 12:17:06.989421 <a moz-do-not-send="true"
href="http://192.168.2.5:5060">192.168.2.5:5060</a> ->
<a moz-do-not-send="true" href="http://108.59.2.133:5060">108.59.2.133:5060</a></div>
<div>SIP/2.0 404 Not here.</div>
<div>To: "1001" <<a moz-do-not-send="true"
href="mailto:sip%3A1001@70.10.163.44">sip:1001@70.10.163.44</a>>;tag=as4b40d9b4.</div>
<div>From: <<a moz-do-not-send="true"
href="mailto:sip%3A001110215178342008@sbc.voxbeam.com">sip:001110215178342008@sbc.voxbeam.com</a>>;tag=3574513019-870807.</div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060">705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060</a>.</div>
<div>CSeq: 2 BYE.</div>
<div>Via: SIP/2.0/UDP
108.59.2.133;received=108.59.2.133;rport=5060;branch=z9hG4bK2deb.8bfd0b06.0.</div>
<div>Content-Length: 0.</div>
<div><br>
</div>
<div><br>
</div>
<div style="">Or is asterisk the culprit? Looking at the
forwarded INVITE (on the asterisk server), I see that the RR
has been re-written, as opposed to appended when contacting
the provider:</div>
<div style=""><br>
</div>
<div style=""><br>
</div>
<div style="">
<div>U 2013/04/09 12:52:52.109611 <a moz-do-not-send="true"
href="http://192.168.2.10:5060">192.168.2.10:5060</a>
-> <a moz-do-not-send="true"
href="http://108.59.2.133:5060">108.59.2.133:5060</a></div>
<div>INVITE <a moz-do-not-send="true"
href="mailto:sip%3A001110215178342008@sbc.voxbeam.com">sip:001110215178342008@sbc.voxbeam.com</a>
SIP/2.0.</div>
<div>Via:
SIP/2.0/UDP 70.10.163.44:5060;branch=z9hG4bK75a764b9;rport.</div>
<div>Max-Forwards: 70.</div>
<div>From: "1001" <<a moz-do-not-send="true"
href="mailto:sip%3A1001@70.10.163.44">sip:1001@70.10.163.44</a>>;tag=as234a7f7d.</div>
<div>To: <<a moz-do-not-send="true"
href="mailto:sip%3A001110215178342008@sbc.voxbeam.com">sip:001110215178342008@sbc.voxbeam.com</a>>.</div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:1001@70.10.163.44:5060">sip:1001@70.10.163.44:5060</a>>.</div>
<div>Call-ID: <a moz-do-not-send="true"
href="http://5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060">5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060</a>.</div>
<div>CSeq: 102 INVITE.</div>
<div>User-Agent: Asterisk PBX
UNKNOWN__and_probably_unsupported.</div>
<div>Date: Tue, 09 Apr 2013 16:52:52 GMT.</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH.</div>
<div>Supported: replaces, timer.</div>
<div>Content-Type: application/sdp.</div>
<div>
Content-Length: 310.</div>
<div>.</div>
<div>v=0.</div>
<div>o=root 731333659 731333659 IN IP4 70.10.163.44.</div>
<div>s=Asterisk PBX UNKNOWN__and_probably_unsupported.</div>
<div>c=IN IP4 70.10.163.44.</div>
<div>t=0 0.</div>
<div>m=audio 30434 RTP/AVP 18 101.</div>
<div>a=rtpmap:18 G729/8000.</div>
<div>a=fmtp:18 annexb=no.</div>
<div>a=rtpmap:101 telephone-event/8000.</div>
<div>a=fmtp:101 0-16.</div>
<div>a=silenceSupp:off - - - -.</div>
<div>a=ptime:20.</div>
<div>a=sendrecv.</div>
<div><br>
</div>
<div><br>
</div>
<div style="">Can we get an externally initiated BYE working
in an OpenSIPS->Asterisk integration? If so, some
suggestions would be appreciated. Maybe just really the
non-loose route BYE to asterisk?</div>
<div style="">Is adding topology hiding functionality
a cumbersome task...</div>
<div style=""><br>
</div>
<div style="">Thanks in Advance,</div>
<div style=""><br>
</div>
<div style="">N.</div>
<div><br>
</div>
</div>
</div>
</div>
<pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
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</pre>
</blockquote>
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